> On Jun 17, 2024, at 5:52 PM, Rémi Denis-Courmont <r...@remlab.net> wrote:
>
>
>
> Le 17 juin 2024 13:18:11 GMT+02:00, Yigithan Yigit
> <yigithanyigitde...@gmail.com <mailto:yigithanyigitde...@gmail.com>> a écrit :
>> ---
>> libavfilter/af_volumedetect.c | 159 ++++++++++++++++++++++++++++------
>> 1 file changed, 133 insertions(+), 26 deletions(-)
>>
>> diff --git a/libavfilter/af_volumedetect.c b/libavfilter/af_volumedetect.c
>> index 327801a7f9..dbbcd037a5 100644
>> --- a/libavfilter/af_volumedetect.c
>> +++ b/libavfilter/af_volumedetect.c
>> @@ -20,27 +20,51 @@
>>
>> #include "libavutil/channel_layout.h"
>> #include "libavutil/avassert.h"
>> +#include "libavutil/mem.h"
>> #include "audio.h"
>> #include "avfilter.h"
>> #include "internal.h"
>>
>> +#define MAX_DB_FLT 1024
>> #define MAX_DB 91
>> +#define HISTOGRAM_SIZE 0x10000
>> +#define HISTOGRAM_SIZE_FLT (MAX_DB_FLT*2)
>>
>> typedef struct VolDetectContext {
>> - /**
>> - * Number of samples at each PCM value.
>> - * histogram[0x8000 + i] is the number of samples at value i.
>> - * The extra element is there for symmetry.
>> - */
>> - uint64_t histogram[0x10001];
>> + uint64_t* histogram; ///< for integer number of samples at each PCM
>> value, for float number of samples at each dB
>> + uint64_t nb_samples; ///< number of samples
>> + double sum2; ///< sum of the squares of the samples
>> + double max; ///< maximum sample value
>> + int is_float; ///< true if the input is in floating point
>> } VolDetectContext;
>>
>> -static inline double logdb(uint64_t v)
>> +static inline double logdb(double v, enum AVSampleFormat sample_fmt)
>> {
>> - double d = v / (double)(0x8000 * 0x8000);
>> - if (!v)
>> - return MAX_DB;
>> - return -log10(d) * 10;
>> + if (sample_fmt == AV_SAMPLE_FMT_FLT) {
>> + if (!v)
>> + return MAX_DB_FLT;
>> + return -log10(v) * 10;
>> + } else {
>> + double d = v / (double)(0x8000 * 0x8000);
>> + if (!v)
>> + return MAX_DB;
>> + return -log10(d) * 10;
>> + }
>> +}
>> +
>> +static void update_float_stats(VolDetectContext *vd, float *audio_data)
>> +{
>> + double sample;
>> + int idx;
>> + if(!isnormal(*audio_data))
>> + return;
>
> Do we really need to classify floats here? That's probably going to hurt
> perfs badly, and makes an otherwise very vectorisable function not so easily
> vectored.
You could be correct, we might consider checking NaN, Inf+/- values. Otherwise
there is a risk of a crash if someone uses something like this
“aelvalsrc=3.4e39”.
>
>> + sample = fabsf(*audio_data);
>> + if (sample > vd->max)
>> + vd->max = sample;
>> + vd->sum2 += sample * sample;
>> + idx = lrintf(floorf(logdb(sample * sample, AV_SAMPLE_FMT_FLT))) +
>> MAX_DB_FLT;
>
> You're recomputing the same value again, and you seem to be rounding twice in
> a row?
I missed, fixed.
>
>> + vd->histogram[idx]++;
>> + vd->nb_samples++;
>> }
>>
>> static int filter_frame(AVFilterLink *inlink, AVFrame *samples)
>> @@ -51,18 +75,41 @@ static int filter_frame(AVFilterLink *inlink, AVFrame
>> *samples)
>> int nb_channels = samples->ch_layout.nb_channels;
>> int nb_planes = nb_channels;
>> int plane, i;
>> - int16_t *pcm;
>> + int planar = 0;
>>
>> - if (!av_sample_fmt_is_planar(samples->format)) {
>> - nb_samples *= nb_channels;
>> + planar = av_sample_fmt_is_planar(samples->format);
>> + if (!planar)
>> nb_planes = 1;
>> + if (vd->is_float) {
>> + float *audio_data;
>> + for (plane = 0; plane < nb_planes; plane++) {
>> + audio_data = (float *)samples->extended_data[plane];
>> + for (i = 0; i < nb_samples; i++) {
>> + if (planar) {
>> + update_float_stats(vd, &audio_data[i]);
>> + } else {
>> + for (int j = 0; j < nb_channels; j++)
>> + update_float_stats(vd, &audio_data[i * nb_channels
>> + j]);
>> + }
>> + }
>> + }
>> + } else {
>> + int16_t *pcm;
>> + for (plane = 0; plane < nb_planes; plane++) {
>> + pcm = (int16_t *)samples->extended_data[plane];
>> + for (i = 0; i < nb_samples; i++) {
>> + if (planar) {
>> + vd->histogram[pcm[i] + 0x8000]++;
>> + vd->nb_samples++;
>> + } else {
>> + for (int j = 0; j < nb_channels; j++) {
>> + vd->histogram[pcm[i * nb_channels + j] + 0x8000]++;
>> + vd->nb_samples++;
>> + }
>> + }
>> + }
>> + }
>
> Can't we pick the correct implementation (planar/packed and float/int) once
> and for all whilst configuring the filter?
Actually sounds good, I am going to try.
Thanks for the feedback!
>
>> }
>> - for (plane = 0; plane < nb_planes; plane++) {
>> - pcm = (int16_t *)samples->extended_data[plane];
>> - for (i = 0; i < nb_samples; i++)
>> - vd->histogram[pcm[i] + 0x8000]++;
>> - }
>> -
>> return ff_filter_frame(inlink->dst->outputs[0], samples);
>> }
>>
>> @@ -73,6 +120,20 @@ static void print_stats(AVFilterContext *ctx)
>> uint64_t nb_samples = 0, power = 0, nb_samples_shift = 0, sum = 0;
>> uint64_t histdb[MAX_DB + 1] = { 0 };
>>
>> + if (!vd->nb_samples)
>> + return;
>> + if (vd->is_float) {
>> + av_log(ctx, AV_LOG_INFO, "n_samples: %" PRId64 "\n",
>> vd->nb_samples);
>> + av_log(ctx, AV_LOG_INFO, "mean_volume: %.1f dB\n", -logdb(vd->sum2
>> / vd->nb_samples, AV_SAMPLE_FMT_FLT));
>> + av_log(ctx, AV_LOG_INFO, "max_volume: %.1f dB\n",
>> -2.0*logdb(vd->max, AV_SAMPLE_FMT_FLT));
>> + for (i = 0; i < HISTOGRAM_SIZE_FLT && !vd->histogram[i]; i++);
>> + for (; i >= 0 && sum < vd->nb_samples / 1000; i++) {
>> + if (!vd->histogram[i])
>> + continue;
>> + av_log(ctx, AV_LOG_INFO, "histogram_%ddb: %" PRId64 "\n",
>> MAX_DB_FLT - i, vd->histogram[i]);
>> + sum += vd->histogram[i];
>> + }
>> + } else {
>> for (i = 0; i < 0x10000; i++)
>> nb_samples += vd->histogram[i];
>> av_log(ctx, AV_LOG_INFO, "n_samples: %"PRId64"\n", nb_samples);
>> @@ -92,26 +153,61 @@ static void print_stats(AVFilterContext *ctx)
>> return;
>> power = (power + nb_samples_shift / 2) / nb_samples_shift;
>> av_assert0(power <= 0x8000 * 0x8000);
>> - av_log(ctx, AV_LOG_INFO, "mean_volume: %.1f dB\n", -logdb(power));
>> + av_log(ctx, AV_LOG_INFO, "mean_volume: %.1f dB\n",
>> -logdb((double)power, AV_SAMPLE_FMT_S16));
>>
>> max_volume = 0x8000;
>> while (max_volume > 0 && !vd->histogram[0x8000 + max_volume] &&
>> !vd->histogram[0x8000 - max_volume])
>> max_volume--;
>> - av_log(ctx, AV_LOG_INFO, "max_volume: %.1f dB\n", -logdb(max_volume *
>> max_volume));
>> + av_log(ctx, AV_LOG_INFO, "max_volume: %.1f dB\n",
>> -logdb((double)(max_volume * max_volume), AV_SAMPLE_FMT_S16));
>>
>> for (i = 0; i < 0x10000; i++)
>> - histdb[(int)logdb((i - 0x8000) * (i - 0x8000))] += vd->histogram[i];
>> + histdb[(int)logdb((double)(i - 0x8000) * (i - 0x8000),
>> AV_SAMPLE_FMT_S16)] += vd->histogram[i];
>> for (i = 0; i <= MAX_DB && !histdb[i]; i++);
>> for (; i <= MAX_DB && sum < nb_samples / 1000; i++) {
>> - av_log(ctx, AV_LOG_INFO, "histogram_%ddb: %"PRId64"\n", i,
>> histdb[i]);
>> + av_log(ctx, AV_LOG_INFO, "histogram_%ddb: %"PRId64"\n", -i,
>> histdb[i]);
>> sum += histdb[i];
>> }
>> + }
>> +}
>> +
>> +static int config_output(AVFilterLink *outlink)
>> +{
>> + AVFilterContext *ctx = outlink->src;
>> + VolDetectContext *vd = ctx->priv;
>> + size_t histogram_size;
>> +
>> + vd->is_float = outlink->format == AV_SAMPLE_FMT_FLT ||
>> + outlink->format == AV_SAMPLE_FMT_FLTP;
>> +
>> + if (!vd->is_float) {
>> + /*
>> + * Number of samples at each PCM value.
>> + * Only used for integer formats.
>> + * For 16 bit signed PCM there are 65536.
>> + * histogram[0x8000 + i] is the number of samples at value i.
>> + * The extra element is there for symmetry.
>> + */
>> + histogram_size = HISTOGRAM_SIZE + 1;
>> + } else {
>> + /*
>> + * The histogram is used to store the number of samples at each dB
>> + * instead of the number of samples at each PCM value.
>> + */
>> + histogram_size = HISTOGRAM_SIZE_FLT + 1;
>> + }
>> + vd->histogram = av_calloc(histogram_size, sizeof(uint64_t));
>> + if (!vd->histogram)
>> + return AVERROR(ENOMEM);
>> + return 0;
>> }
>>
>> static av_cold void uninit(AVFilterContext *ctx)
>> {
>> + VolDetectContext *vd = ctx->priv;
>> print_stats(ctx);
>> + if (vd->histogram)
>> + av_freep(&vd->histogram);
>> }
>>
>> static const AVFilterPad volumedetect_inputs[] = {
>> @@ -122,6 +218,14 @@ static const AVFilterPad volumedetect_inputs[] = {
>> },
>> };
>>
>> +static const AVFilterPad volumedetect_outputs[] = {
>> + {
>> + .name = "default",
>> + .type = AVMEDIA_TYPE_AUDIO,
>> + .config_props = config_output,
>> + },
>> +};
>> +
>> const AVFilter ff_af_volumedetect = {
>> .name = "volumedetect",
>> .description = NULL_IF_CONFIG_SMALL("Detect audio volume."),
>> @@ -129,6 +233,9 @@ const AVFilter ff_af_volumedetect = {
>> .uninit = uninit,
>> .flags = AVFILTER_FLAG_METADATA_ONLY,
>> FILTER_INPUTS(volumedetect_inputs),
>> - FILTER_OUTPUTS(ff_audio_default_filterpad),
>> - FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P),
>> + FILTER_OUTPUTS(volumedetect_outputs),
>> + FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_S16,
>> + AV_SAMPLE_FMT_S16P,
>> + AV_SAMPLE_FMT_FLT,
>> + AV_SAMPLE_FMT_FLTP),
>> };
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